similar to: [asterisk-app-dev] ARI Get Channel Variable

On Sat, Jun 17, 2023 at 2:55 PM TTT <lists at telium.io> wrote: > Based on postings it should be possible to get the SIP Call-ID header > value from the ARI. At what point is this value available ? As well, how > do I retrieve that value – something like > > > > GET /channels/{channelId}/pjsip_header?key=Call-Id > > > > But that doesn’t work. >

I tried GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id) But it responds with "message": "Channel not in Stasis application" Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan, so the PJSIP_HEADER function won’t work. So it looks like I’ll

Based on postings it should be possible to get the SIP Call-ID header value from the ARI. At what point is this value available ? As well, how do I retrieve that value - something like GET /channels/{channelId}/pjsip_header?key=Call-Id But that doesn't work. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, Jun 26, 2023 at 4:04 PM TTT <lists at telium.io> wrote: > It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the > entire SIP header for a channel. I also read (on stackoverflow) that the > PJSIP_HEADER function will only return the headers from the INVITE of the > *inbound* channel. > > > > If that’s correct, how would I get the headers from

I think that’s getting me close. I’m trying to get (or recreate) the FROM and TO lines of the header, from a system running PJSIP. I think if I use CHANNEL to get local_uri and local_tag I can recreate a FROM line like: FROM=<URI>;tag=TAG And if I use CHANNEL to get remote_uri and remote_tag I can recreate a FROM line like: TO=<URI>;tag=TAG Would it be correct to assume

Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] >> >> [TOOTAiAudio] >> ; >> ; Call our gateway >> >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) >>  same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) >>  same = n,Return >> >> exten = h,1,NoOp()

It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVITE ? Or will PJSIP_HEADERS() in fact return the

Hi list, to manage an external queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set extra header. Dialplan to call external agent is this one with (Gosub): [TOOTAiAudio] ; ; Call our gateway exten =

>> There are SOME protocol level things accessible using CHANNEL[1] but that's it. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL I am trying to use the CHANNEL function listed above from the AMI. Since it is not an AMI “action”, but rather a dialplan “function”, I’m trying to figure out how to call this from the AMI. Using a telnet session

The uppercase command made a difference. I now get a call-id as show below. However, does the call-id look valid? The @0.0.0.0 seems strange. action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,call-id) Response: Success ActionID: act1 Variable: CHANNEL(pjsip,call-id) Value: 4decf884e3ae74595906283a74f7154e at 0.0.0.0 As well,

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)

On Sun, Jul 2, 2023 at 4:18 PM TTT <lists at telium.io> wrote: > >> There are SOME protocol level things accessible using CHANNEL[1] but > that's it. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL > > > > > > I am trying to use the CHANNEL function listed above from the AMI. Since > it is not an AMI

Hi all, I’m trying to rewrite Diversion header when call forwarding is done on the phone. The phone sends "302 Moved Temporarily" response and sets Diversion header to a local number, but before Asterisk sends this call towards TSP provider I need to change Diversion header to a full PSTN number. I am using PJSIP_HEADER in a pre-dial handler (configuration is below). On the same

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message. I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each

I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from

Hi. The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too. So, can I use PJSIP_HEADER to read the SIP 183 message header? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel

Hi List Implementing screening and routing I have stumbled over this issue: [pbx-router] exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION}) same => n,Set(SOURCE=${CHANNEL(name)}) same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same